CN1149790C - Speech system using speech-carrying protocol of interconnected network protocol as basis - Google Patents
Speech system using speech-carrying protocol of interconnected network protocol as basis Download PDFInfo
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L12/00—Data switching networks
- H04L12/66—Arrangements for connecting between networks having differing types of switching systems, e.g. gateways
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M3/00—Automatic or semi-automatic exchanges
- H04M3/42—Systems providing special services or facilities to subscribers
- H04M3/487—Arrangements for providing information services, e.g. recorded voice services or time announcements
- H04M3/493—Interactive information services, e.g. directory enquiries ; Arrangements therefor, e.g. interactive voice response [IVR] systems or voice portals
- H04M3/4938—Interactive information services, e.g. directory enquiries ; Arrangements therefor, e.g. interactive voice response [IVR] systems or voice portals comprising a voice browser which renders and interprets, e.g. VoiceXML
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L12/00—Data switching networks
- H04L12/64—Hybrid switching systems
- H04L12/6418—Hybrid transport
- H04L2012/6472—Internet
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L12/00—Data switching networks
- H04L12/64—Hybrid switching systems
- H04L12/6418—Hybrid transport
- H04L2012/6481—Speech, voice
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M7/00—Arrangements for interconnection between switching centres
- H04M7/0003—Interconnection between telephone networks and data networks
- H04M7/0006—Interconnection between telephone networks and data networks where voice calls cross both networks
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Abstract
A VoIP-enabled speech server can include a speech application which can be configured to communicate with a VoIP telephony gateway server over a VoIP communications path. The VoIP-enabled speech server can also include a VoIP-compliant call control interface to the VoIP telephony gate server, the VoIP-compliant call control interface establishing the VoIP communications path. In operation, the speech application can receive VoIP-compliant packets from the VoIP telephony gateway server over the VoIP communications path. Subsequently, digitized audio data can be reconstructed from the VoIP-compliant packets, and the digitized audio data can be speech-to-text converted. Additionally, text can be synthesized into digitized audio data and the digitized audio data can be encapsulated in VoIP-compliant packets which can be transmitted over the VoIP communications path to the telephony gateway server.
Description
Technical field
The present invention relates to field of speech recognition, relate in particular to the voice application in the protocol network that is used in Voice over IP.
Background technology
The LAN phone, i.e. " phone that provides by Packet Switch Data Network and data service integrated " is a kind of the individual is risen to a new high degree and corresponding cost is reduced to lower level technology to individual's communication.The LAN phone allows to use how flexible, worthwhile application, for example call distribution automatically, Interactive Voice Response, voice record etc.It and relatively limited integrated (computer-phone is integrated) that existing voice/the aggregation of data pattern provides form contrast, and speech business keeps separating and carrying by circuit-switched link with data service in computer-phone is integrated.In view of the old model use circuit exchanging telephone structure of integrated data and voice is carried out data communication, so there is a following shortcoming: the data service available bandwidth is relatively very low, because data service " burst " characteristic causes the low and limited integrated possibility of voice/data of efficient of the Circuit-switched data communication of use, these significant disadvantages cause producing existing topology, wherein the IP data server bundlees mutually with proprietary PBX or sound circuit switch, so that between the circuit and the packet switching network, provide loose integrated, and voice are net loaded by circuit switching.
The prevailing application of one of them of LAN phone is to use in company's the Internet/intranet environment, and this is called as IP phone.Voice over IP (" VoIP ") agreement is the agreement of support voice business by the IP network transmission.In voip network, the analogue voice signal that receives from simulated voice audio-source (for example PSTN or microphone) is digitized, compresses and convert to the IP grouping so that transmit by IP network.Some well-known agreements realize comprising H.323, conversation initialized protocol (" SIP ") and primary gateway control protocol (" MGCP ") the VoIP protocol specification.
The common application of IP phone is integrated speech mail (" v-mail ") and Email (" e-mail ").Another application comprises the voice record that finance or emergency response tissue carry out.In addition, also help automatic call distribution (ACD), the ACD server can be finished whereby based on the call of going into of numerical value and line up.At last, interactive voice response system can wherein respond as workflow component and carry out pre-programmed in server in conjunction with IP phone.However, speech recognition and phonetic synthesis application (" voice application ") still falls behind in IP phone is used.
Especially, at the voice application of real-time audio signal operation, be impatient at the relevant stand-by period of conventional data communications.Like this, the occasion in the combined IP phone topology of voice application, voice application and IP phone server are combined closely so that get rid of the negative effect of based on network time delay in advance.Therefore, design and develop such IP phone voice application can be in the same place with the proprietary characteristic close-coupled of IP phone server.
Close-coupled between voice application and the IP phone server has limited the design and the expansion of voice application in fact.Especially, in existing pattern, the voice application design must comprise and directly relates to voice data transmission that selected agreement will break into grouping to speech recognition system and from the function of speech synthesis system transmission speech data.The development of advanced speech host-host protocol (being the close-coupled of IP phone server and voice application in essence) can force voice application to design again.Thereby this just needs the voice system based on VoIP, and wherein the design of voice application will keep separating with realizing with realizing with the design of IP telephony system.
Summary of the invention
An object of the present invention is to provide voice system, wherein keep the design of voice application and the design and the realization of realization and IP telephony system to be separated based on VoIP.Further purpose of the present invention provides a voice server that starts VoIP, and it can receive from the audio frequency of IP telephony system by the VoIP net and import.Another object of the present invention provides a kind of method, in the VoIP net voice application is connected to caller gateway server.At last, the purpose of this invention is to provide voice server and the method for each voice system, startup VoIP, be used for voice application being connected to caller gateway server by the measured interface that uses VoIP net, caller gateway server and voice application based on VoIP.
These and other purposes of the present invention realize that in the voice system based on VoIP this system comprises: the voip phone gateway server; At least one voice server, each voice server comprise a voice application that starts VoIP; The calling control interface of deferring to VoIP between voip phone gateway server and voice server; With, a VoIP communication path between the voice application in voip phone gateway server and at least one voice server.In the voice system based on VoIP, voip phone gateway server and voice application can be set up meaning VoIP communication path by the calling control interface of abiding by VoIP.
Be in operation, the voip phone gateway server can be from the telephony interface received audio signal, audio signal digitizing is become digital audio data, digital audio data is compressed into the grouping of deferring to VoIP, and the grouping of using the VoIP agreement will defer to VoIP by the VoIP communication path is sent to the voice application at least one voice server.Accordingly, voice application can receive the grouping of deferring to VoIP, according to the grouping reconstructed number voice data of deferring to VoIP, and digital audio data is carried out voice-to the conversion of-text.And voice application can be synthesized to text in the digital audio data, encapsulates digital audio data in the grouping of deferring to VoIP, and the grouping that will defer to VoIP is sent to the voip phone gateway server by the VoIP communication path.Then, the voip phone gateway server receives the grouping of deferring to VoIP, transmits this digital audio data according to the grouping reconstructed number voice data of deferring to VoIP and by telephony interface.
In one aspect of the invention, the voip phone server can comprise that a telephony interface and VoIP close and keeps.VoIP closes and keeps and can receive voice call by telephony interface, and this is responded, and VoIP closes to keep and can select a voice server from voice server.In case voice server is chosen, VoIP closes the voice application of keeping the startup VoIP in the selected voice server of notice: received voice call.
In another aspect of this invention, voice server can comprise speech recognition engine; Text-to-speech engine; Be used for setting up the calling control interface that voice call connects by the voip phone gateway server; With, audio data path.Notice that audio data path can be sent to speech recognition engine with voice data from the data flow mode by the voice call connection of having set up.Similarly, audio data path can connect from text-voice data is sent out in the data flow mode to-speech engine by the voice call set up.
In another aspect of this invention, voice application can be a voice browser.Voice browser can be fetched web content in response to the voice commands of receiving on the VoIP communication path.And voice browser can carry out the web content of fetching phonetic synthesis and become voice data.At last, voice browser can be sent to the voip phone gateway server by the VoIP communication path with voice data.It should be noted that web content can be speech XML (extend markup language) file.
Preferably, the voice server measured interface that can use voip phone gateway server, VoIP communication path and voice application is realized.Especially, voice server can comprise speech recognition engine; Text-to-speech engine; JSAPI (java speech application interface) speech interface; A JTAPI (java Telephony Application Programming Interface) telephony interface; With a JMF (java media framework) media interface.The JTAPI telephony interface can be set up a voice call and connect, so that transmit digital audio-frequency data between caller gateway server and voice application.The JMF media interface can be set up data path, so that transmit digital audio-frequency data between voice application and voice call connection.The JSAPI speech interface can be sent to speech recognition engine from voice application with digital audio data.Similar, the JSAPI speech interface can be with the phonetic synthesis voice data from text-be sent to voice application to-speech engine.
The present invention can also realize in starting the voice server of VoIP, and voice server comprises and can be configured so that the voice application of communicating by letter with the voip phone gateway server by the VoIP communication path.The voice server that starts VoIP also covers the calling control interface of deferring to VoIP of voip phone gateway server, and this calling control interface of deferring to VoIP is set up the VoIP communication path.Be in operation, voice application can receive the grouping of deferring to VoIP by the VoIP communication path from the voip phone gateway server.Then, according to the grouping reconstructed number voice data of deferring to VoIP, then digital audio data is carried out voice-to the conversion of-text.And text can be synthesized in the digital audio data, and digital audio data is packaged into and can be sent to grouping caller gateway server, that defer to VoIP by the VoIP communication path.
The voice server that starts VoIP on the other hand in, the voice server that starts VoIP comprise speech recognition engine, text-to-speech engine and audio data path.Audio data path can be sent to speech recognition engine with voice data in the data flow mode by the voice call connection of having set up.And audio data path can connect from text-voice data is sent out with stream mode to-speech engine by the voice call set up.
Preferably, voice application is a voice browser.Voice browser can be fetched web content in response to the voice commands of receiving on the VoIP communication path.And voice browser can carry out the web content of fetching phonetic synthesis and become voice data.Then, voice browser can be sent to the voip phone gateway server by the VoIP communication path with voice data.It should be noted that web content can be a speech XML file.
Preferably, the voice server that starts VoIP can be realized by the measured interface that uses voip phone gateway server, VoIP communication path and voice application.Especially, the voice server that starts VoIP comprises the JTAPI telephony interface, so that set up a voice call connection that is used to transmit digital audio-frequency data between caller gateway server and voice application.And the voice server that starts VoIP also comprises the JMF media interface, so that set up a data path that is used to transmit digital audio-frequency data between voice application and voice call connection.At last, the voice server that starts VoIP also comprises the JSAPI speech interface, so that digital audio data can be sent to speech recognition engine from voice application, and can be with the voice data of phonetic synthesis from text-be sent to voice application to-speech engine.
At last, the present invention comprises a kind of method, and it can be connected to voice application on the caller gateway server in the VoIP net.The method comprises step: set up a VoIP communication path and configured voice with the voip phone gateway server and use so that communicate by VoIP communication path and the caller gateway server of having set up.And, receive the grouping of deferring to VoIP from caller gateway server by the VoIP communication path of having set up.According to the grouping reconstructed number voice data of deferring to VoIP, then, digital audio data is carried out voice-to the conversion of-text.And this method comprises step: text is synthesized to digital audio data; Digital audio data is packaged into the grouping of deferring to VoIP; And the grouping that will defer to VoIP by the VoIP communication path is sent to caller gateway server.
In preferred embodiments, the method further comprises step: fetch web content in response to the speech recognition voice commands of receiving on the VoIP communication path; The web content of fetching is synthesized voice data; Voice data is sent to caller gateway server by the VoIP communication path.It should be noted that web content can be a speech XML file.
Description of drawings
Shown existing preferred embodiment in the accompanying drawings, be appreciated that the present invention is not limited to accurate scheme and the means that show here.
Fig. 1 graphic extension is according to the voice system based on VoIP of the present invention.
Fig. 2 is the preferred structure figure of voip phone gateway server among Fig. 1.
Fig. 3 is the preferred structure figure of voice server among Fig. 1.
Embodiment
The present invention is a voice system based on Voice over IP (" VoIP "), and wherein voice server can be connected to the caller gateway server in the VoIP net.Caller gateway server can receive voice call from external telephone network, public switched telephone network (" PSTN ") for example, integrated services digital network (" ISDN ") or the like.Voice server comprises can be by the voice application of VoIP communication path reception from the real-time speech input of the voice call of caller gateway server generation.Similar, voice application can be sent to caller gateway server with the phonetic synthesis voice data by the VoIP communication path, and finally arrives the termination point of external telephone network.Merit attention, voice application can receive the voice browser instruction by voice call, and as its response, voice application is fetched web content from external web server.And web content can carry out phonetic synthesis and also transmit by the VoIP communication path, as the part of voice call.In preferred embodiments, web content can be speech XML file.
Fig. 1 explanation is according to the voice system based on VoIP of preferred embodiment.For example H.323, SIP and MGCP realize notice that well known in the art, the VoIP standard can be used some Open Standards.But H.323 the present invention adopts, though the present invention is not confined to this VoIP specific implementation.As shown in Figure 1, be in operation, the user can use telephone plant 1 to initiate voice call.Voice call is attempted linking to each other with voip phone gateway server 3 by telephone network 2 (for example PSTN or ISDN).Voip phone gateway server 3 becomes to be built in the IP address of the equipment in the VoIP net 4 with the address transition of the intended recipinent of voice call, and this equipment is for starting the voice server 5 of VoIP in this example.Then, voip phone gateway server 3 starts the voice server 5 acceptable voice calls of VoIP to voice server 5 notices that start VoIP.After receiving voice call, the voice server 5 that starts VoIP is set up the VoIP communication path between the voice server 5 of voip phone gateway server 3 and startup VoIP, the grouping of deferring to VoIP of voice data can transmission between the voice server 5 of voip phone gateway server 3 and startup VoIP like this.Under this mode, the voice data that is derived from telephone plant 1 can be received and handle in the voice server 5 that starts VoIP.Equally, the voice data that is derived from the voice server 5 that starts VoIP can send back telephone plant 1.
Especially, in preferred embodiments, the voice server 5 that starts VoIP can receive voice commands from telephone plant 1 so that fetch web content from the Web server 7 of data communication network 6.Web content 8 can be speech XML file 8 particularly.As response, the voice server 5 that starts VoIP can be fetched speech XML file 8 also according to the indication Composite tone data that are included in the speech XML file 8 from Web server 7.Then, He Cheng voice data is sent to voip phone gateway server 3 and is ultimately delivered to telephone plant 1 by VoIP net 4.
Therefore, after telephony interface 11 was received voice call, calling control can be sent to the VoIP pass by calling control interface 13 and keep 14.Especially, VoIP close keep 14 can be by other component communications of data path 17 and voip phone gateway server 3.And, call out control interface 13 and can be included in VoIP and close and keep in 14, so that control is closed the foundation of keeping 14 voice calls of handling, carries out and stopped by VoIP.Because the preferred realization of VoIP is based on the realization of the H.323 standard of RTP, so calling control interface 13 is based on calling control interface H.323.
Then, this control is sent to the VoIP pass and keeps 14, the voice server 5 of the startup VoIP of the most suitable reception voice call in the state of the voice server 5 of each startup VoIP and the definite VoIP net 4 in the call processing device 16 check VoIP nets 4 of use advanced call management function 15.As a result, the VoIP pass is kept the voice server 5 of a suitable startup VoIP of selection and is notified voice calls to the voice server 5 of this selected startup VoIP.
After receiving notice, the voice server 5 of selected startup VoIP is set up a VoIP communication path 18 between the voice server 5 of voip phone gateway server 3 and startup VoIP, transmit the grouping of deferring to VoIP by this path.Then, telephony interface 11 will be included in audio signal in the voice call and be digitized into and be digital audio data, this digital audio data is compressed into the grouping of deferring to VoIP and use the VoIP agreement grouping of deferring to VoIP to be sent to the voice server 5 of selected startup VoIP by VoIP communication path 18.
Noticeable, the present invention is not limited to the ad hoc arrangement of voip phone gateway server 3.Especially, VoIP is closed keep 14 corpus separatums that are described as being independent of voip phone gateway server 3 remaining components and do not mean and to limit the present invention like this.But VoIP closes and to keep 14 placement just in order to reach illustrative purposes among Fig. 2.And the present invention closes about voip phone gateway server 3 and VoIP that to keep the scope that combines only be that VoIP closes and keeps that 14 terminals that can be in VoIP net 4 be suitable for this voice call by identification are carried out call manager and restricted because require voip phone gateway server 3 can receive voice call.
Start the preferred structure of the voice server 5 of VoIP in Fig. 3 key diagram 1.The voice server 5 that starts VoIP can be realized by enough general networks servers, the general networks server comprises CPU (CPU) and internal storage device traditionally, for example random-access memory (ram) 21 and read-only storage 22 are as hard disk drive (HDD).Support voice operation because start the voice server 5 of VoIP, provide the Audio Processing ability so that be provided as the voice server 5 that starts VoIP so the voice server 5 of startup VoIP also comprises the voicefrequency circuit (not shown).
The voice server 5 that starts VoIP can be carried out various application programs on operating system 23 in read-only storage 22 storage operating systems 23.And read-only storage 22 can storaged voice application 24 and voip phone module 25.Operating system 23 comprises arbitrary suitable operating system, for example the Windows NT of Microsoft
, Sun Solaris
Or DebianLinux.Noticeable, the present invention is unrestricted on the relevant problem of the arrangement of voice application 24 and voip phone module 25 and operating system 23.But each can integrate with various combining forms and another.For example, voip phone module 25 can be integrated into operating system 23.Perhaps, the independence of 25 maintenances of voip phone module and operating system 23.
And the present invention is not subject to the memory location of voip phone module 25, voice application 24 and its assembly.But, the present invention can more realize in the complicated distributed system, wherein each assembly resides in a plurality of webservers and in far apart process address and carries out, and each application communicates by well-known interprocedual communication mechanism, for example TCP/IP.When the voice server 5 that starts VoIP was booted, operating system 23 was loaded onto among the RAM21.Then, voice application 24 and voip phone module 25 are loaded onto among the RAM21 and carry out therein.In case carry out, the voice server 5 that starts VoIP just is configured to receive voice call and follow-up data by the VoIP communication path.
Voice application 24 comprises speech recognition engine 34 and text-to-speech engine 35.Be in operation, the voice server 5 that starts VoIP can receive the grouping of deferring to VoIP, and according to the grouping reconstructed number voice data of deferring to VoIP, and in speech recognition engine 34 digital audio data is carried out speech-to the conversion of-text.On the contrary, voice application 24 can be at text-in-speech engine 35 text is synthesized digital audio data, digital audio data is encapsulated in the grouping of deferring to VoIP, and these groupings of deferring to VoIP is sent to voip phone gateway server 3 by VoIP communication path 18.
In the preferred embodiment, voice application comprises voice browser 30.Noticeable, voice browser 30 can be fetched web content in response to the voice commands of receiving on the VoIP communication path 18, carries out speech-to the conversion of-text by speech recognition engine 34, and by carrying out decipher in the voice browser 30.And voice browser 30 is sent to text-to-speech engine 35 with the web content that receives so that by VoIP communication path 18 audio data transmission of phonetic synthesis is being synthesized to the voip phone gateway server 3 lang sound that advances.
Preferably, voice application can use measured, realize to the interface of VoIP communication, speech recognition and speech-sound synthesizing function.Especially, voice application 24 is included in speech recognition and text-to the JSAPI speech interface 33 between-speech engine 34,35 and the voice browser 30.And voice application 24 also is included in the JTAPI telephony interface 31 between phone module 25 and the voice browser 30.At last, voice application 24 also is included in the JMF media interface 32 between phone module 25 and the voice browser 30.
Voice browser 30 uses JTAPI telephony interface 31 to set up the voice call connection so that transmit the grouping of deferring to VoIP that comprises digital audio-frequency data between caller gateway server 3 and voice application 24.JMF media interface 32 is set up VoIP communication data path so that transmit the grouping of deferring to VoIP that comprises digital audio-frequency data between voice application 24 and voice call connection.JSAPI speech interface 33 can pass to speech recognition engine 34 from voice application 24 with digital audio data.Similar, JSAPI speech interface 33 can be with the voice data of phonetic synthesis from text-pass to voice application 24 to-speech engine 35.
Voice system based on VoIP of the present invention allows the user to use telephone interview Web website.The user can hear the voice suggestion of describing Web website and characteristic thereof.Like this, the presenting of voice system based on VoIP just is similar to the integrated voice responding system.In the demonstration of Web website, the user can provide spoken command to the voice system based on VoIP so that select option and input information is finished based on the form of Web etc.Voice system based on VoIP can be fetched the web content that comprises as the speech XML of its webpage descriptive language easily.
Significant element of the present invention is a voice server, and it can use well-known standard VoIP agreement H.323 agreement transmission and reception audio frequency and control messages.Use the VoIP agreement to allow other unit of voice server and voice system to keep isolating, therefore allow voice server to be optimized better.Use independent voice server also to simplify development process significantly, because it makes voice server from carrying out freeing mutual mouthful the details with one of a large amount of available phone hardware devices.At last, use the voice server of independent startup VoIP to allow voice server not needing under the special development situation, with the standard of support H.323 or the arbitrary voip phone system of other VoIP agreements use jointly.
According to the present invention program's voice system based on VoIP have with voice server from the caller gateway server of telephone network interface telephony hardware and software detail in the advantage that isolates.And the preferred embodiments of the invention combine standard agreement and are used for transmission of audio and call control information and give voice server, have so just simplified and optimized the design of homotype product.In addition, the combined standard agreement is used for transmission of audio and call control information allows voice server to operate telephony hardware and the software of supporting standard VoIP agreement widely to voice server.
The present invention also comprises voice application is connected to method on the voip phone gateway server.The mode of the enough hardware of the method energy, software or the two combination realizes.Can realize with centralized system in a computer system that according to machine readable memory of the present invention perhaps the distribution mode that intersperses among in some computer systems that interconnect each other with different units realizes.Being suitable for of any kind of realizes that the computer system of method described herein or other devices all can accept.Typical combination of hardware can be the general purpose department of computer science computer program of unifying, and when described program was loaded and carries out, the control computer system was so that realize method described herein.The present invention also can be embedded in the computer program, and it comprises all characteristics that realize method described herein, and it can realize these methods in the time of in being loaded into computer system.
Computer program herein means and can express one group of instruction with any language, code or symbol, purpose is to make system have information processing capability so that finish specific function, specific function can directly finish or arbitrary below or two steps after finish: (a) convert other language, code or symbol to; (b) reproduce with the different materials form.The present invention disclosed herein can be a kind of method in the embeddeding computer program, and it can be realized by the commercial development instrument that the programmer is used for above-described operating system 23.
Claims (17)
1. based on the voice system of Voice over IP VoIP, comprise:
A voip phone gateway server;
At least one voice server, each voice server comprise a voice application;
Calling control interface of deferring to VoIP between described voip phone gateway server and described voice server; With
, the VoIP communication path between described voip phone gateway server and the described voice application in described at least one voice server;
Described voip phone gateway server and described voice application are set up described VoIP communication path by the calling control interface of deferring to VoIP;
Described voip phone gateway server is from the telephony interface received audio signal, audio signal digitizing is become digital audio data, described digital audio data is compressed into the grouping of deferring to VoIP, and uses a VoIP agreement this grouping of deferring to VoIP to be sent to voice application in described at least one voice server by described VoIP communication path;
Described voice application at least one described voice server receives described grouping of deferring to VoIP, defers to the described digital audio data of grouping reconstruct of VoIP according to this, and this digital audio data is carried out voice-to the conversion of-text;
Described voice application at least one described voice server synthesizes digital audio data with text, in the grouping of deferring to VoIP, encapsulate this digital audio data, and this grouping of deferring to VoIP is sent to the voip phone gateway server by the VoIP communication path;
Described voip phone gateway server receives described grouping of deferring to VoIP, defers to the described digital audio data of grouping reconstruct of VoIP according to this, and transmits this digital audio data by described telephony interface.
2. the voice system based on Voice over IP VoIP of claim 1 explanation is characterized in that wherein said voip phone gateway server comprises:
A telephony interface; With,
A VoIP closes and keeps;
Described VoIP closes to keep by described telephony interface and receives voice call;
Described VoIP closes to keep and select a voice server from described at least one voice server;
Described VoIP closes and keeps the voice application of notifying the startup VoIP in this selected voice server: received described voice call.
3. the voice system based on Voice over IP VoIP of claim 1 explanation is characterized in that wherein said at least one voice server comprises:
A speech recognition engine;
Text-to-speech engine;
One is used for setting up the calling control interface that voice call connects by the voip phone gateway server; With,
An audio data path;
Described audio data path is sent to speech recognition engine with voice data with the data flow form by described voice call connection of having set up;
Described audio data path connects from text-voice data is sent out with the data flow form to-speech engine by the described voice call set up.
4. the voice system based on Voice over IP VoIP of claim 1 explanation, it is characterized in that, wherein said voice application comprises voice browser, described voice browser is fetched web content in response to the voice commands of receiving on the VoIP communication path, described voice browser carries out phonetic synthesis with this web content of fetching and becomes voice data, and described voice browser is sent to described voip phone gateway server by the VoIP communication path with this voice data.
5. the voice system based on Voice over IP VoIP of claim 4 explanation is characterized in that wherein said web content is a speech expandable mark language XML file.
6. the voice system based on Voice over IP VoIP of claim 1 explanation is characterized in that wherein said at least one voice server comprises:
A speech recognition engine;
Text-to-speech engine;
A java speech application interface JSAPI speech interface;
A java Telephony Application Programming Interface JTAPI telephony interface; With,
A java media framework JMF media interface;
Described JTAPI telephony interface is set up a voice call connection that is used to transmit digital audio-frequency data between described voip phone gateway server and described voice application;
Described JMF media interface is set up a data path that is used to transmit described digital audio-frequency data between described voice application and the connection of described voice call;
Described JSAPI speech interface is sent to described speech recognition engine with described digital audio-frequency data from described voice application;
Described JSAPI speech interface with described phonetic synthesis voice data from described text-be sent to described voice application to-speech engine.
7. start the voice server of Voice over IP VoIP, comprising:
A voice application, described voice application are configured to communicate by letter with the voip phone gateway server by the VoIP communication path; With
Calling control interface of deferring to VoIP to described voip phone gateway server, this calling control interface of deferring to VoIP is set up described VoIP communication path;
Described voice application can receive the grouping of deferring to VoIP by described VoIP communication path from described voip phone gateway server, defer to the grouping reconstructed number voice data of VoIP according to this, and this digital audio data is carried out voice-to the conversion of-text.
8. the voice server of the startup Voice over IP VoIP of claim 7 explanation, it is characterized in that, described voice application synthesizes digital audio data with text, and this digital audio data of encapsulation also is sent to described caller gateway server with this grouping of deferring to VoIP by the VoIP communication path in the grouping of deferring to VoIP.
9. the voice server of the startup Voice over IP VoIP of claim 7 explanation is characterized in that, also comprises:
A speech recognition engine;
Text-to-speech engine; With,
An audio data path;
Described audio data path is sent to speech recognition engine with voice data with the data flow form by described voice call connection of having set up;
Described audio data path connects from text-voice data is sent out with the data flow form to-speech engine by the described voice call set up.
10. the voice server of the startup Voice over IP VoIP of claim 7 explanation, it is characterized in that, wherein said voice application comprises a voice browser, described voice browser is fetched web content in response to the voice commands of receiving on the described VoIP communication path, described voice browser carries out phonetic synthesis with the described web content of fetching and becomes voice data, and this voice browser is sent to described voip phone gateway server with voice data by described VoIP communication path.
11. the voice server of the startup Voice over IP VoIP of claim 10 explanation is characterized in that wherein said web content is a speech expandable mark language XML file.
12. the voice server of the startup Voice over IP VoIP of claim 7 explanation, it is characterized in that, the calling control interface of the wherein said VoIP of deferring to is a java Telephony Application Programming Interface JTAPI telephony interface, is used for setting up between described voip phone gateway server and described voice application a voice call connection that is used to transmit digital audio-frequency data.
13. the voice server of the startup Voice over IP VoIP of claim 12 explanation is characterized in that, also comprises:
A speech recognition engine;
Text-to-speech engine;
A java speech application interface JSAPI speech interface; With
A java media framework JMF media interface;
Described JMF media interface is set up a data path that is used to transmit digital audio-frequency data between described voice application and the connection of described voice call;
Described JSAPI speech interface is sent to described speech recognition engine with described digital audio-frequency data from described voice application;
Described JSAPI speech interface with described phonetic synthesis voice data from described text-be sent to described voice application to-speech engine.
14. the method on the caller gateway server that voice application is connected to Voice over IP VoIP in voip network, it comprises step:
Described voice application is set up a VoIP communication path with the voip phone gateway server, disposes described voice application so that communicate by letter with the voip phone gateway server by the VoIP communication path of having set up;
Described voice application receives the grouping of deferring to VoIP by the VoIP communication path of having set up from the voip phone gateway server;
Described voice application is deferred to the described digital audio data of grouping reconstruct of VoIP according to this, and this digital audio data is carried out voice-to the conversion of-text.
15. the method for claim 14 explanation is characterized in that, further comprises step:
Text is synthesized digital audio data;
This digital audio data of encapsulation in the grouping of deferring to VoIP; And
The grouping of deferring to VoIP is sent to the voip phone gateway server by described VoIP communication path.
16. the method for claim 14 explanation is characterized in that, further comprises step:
Fetch web content in response to the speech recognition voice commands of receiving on the described VoIP communication path;
The described web content of fetching is synthesized voice data; And
This voice data is sent to the voip phone gateway server by described VoIP communication path.
17. the method for claim 16 explanation is characterized in that wherein said web content is a speech expandable mark language XML file.
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| US09/596,769 US6654722B1 (en) | 2000-06-19 | 2000-06-19 | Voice over IP protocol based speech system |
| US09/596769 | 2000-06-19 |
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| EP (1) | EP1178658A3 (en) |
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| CN (1) | CN1149790C (en) |
| TW (1) | TW512619B (en) |
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